WiMAX systems are being deployed in novel frequency bands often presenting challenges in the manner of reflections, shadowing, etc. Experts can meet many of these challenges when they choose the initial RF settings in a system, such as antenna orientations, downtilts, and transmit power. Once these settings have been chosen, it is a labor intensive process to improve them. However, significant gains in resource allocation efficiency can be made by doing so. In order to reduce the labor cost, we have developed a tool to automate the process of "optimizing" system RF attributes. This system is currently in field testing. We describe the nature of the optimizer and the results obtained from laboratory testing.

This paper focuses on delay and jitter for voice traffic in an IP based infrastructure for CDMA cellular networks. During soft handoff, a key aspect of CDMA networks, a mobile station (MS) may maintain connections with multiple base station transceivers (BTS). Each connection is called a soft handoff connection. The MS transmits copies of the voice frame to each BTS. The multiple copies are sent by the BTSs to a selector distributor unit (SDU) that selects the 'best' frame based on frame error conditions. The SDU has to wait for all copies to arrive prior to selection and then forwards the voice frame to the public switched telephone network (PSTN) through a circuit gateway. This causes additional delay and jitter for the voice traffic. We propose multiple schemes to reduce voice delays between the BTS and SDU and subsequently the circuit gateway. These schemes are based on reordering the voice frames based on two identifiers available on each voice packet: (1) their network identifier derived from IP addresses and UDP port numbers; (2) the number of soft handoff connections maintained by the MS. For each scheme, we derive analytical results for the delay distribution and use the results for a numerical comparison of the schemes. We also conduct a simulation study to validate the assumptions made in the analytical work.

We introduce a new protocol to improve the capacity of communication links to carry real-time applications that generate a large number of small packets. The key issue in the transport of such applications is the relatively large protocol overhead for each packet. For example, the overhead for RTP/UDP/IP/PPP/HDLC is 12/8/20/7 bytes respectively resulting in a total of 47 bytes. This is considerable for applications with a small payload (10-20 bytes). RTP/UDP/IP header compression (cRTP) mitigates the problem partially by reducing the corresponding protocol overhead from 40 bytes to a best case of 2 bytes. PPPmux provides a further improvement by multiplexing multiple application-level packets into a single PPP frame thereby reducing the PPP overhead per packet. One potential deployment of PPPmux is in cellular networks (IS-95A and UMTS) on links connecting base stations to the radio access network controllers (RNC). The paper provides numerical results listing capacity improvements over existing schemes. Since, multiplexing concatenates multiple IP packets into a single PPP frame, a frame loss would result in loss of multiple IP packets. We provide guidelines on the choosing the optimal multiplexing size that gives the best trade-off between increase in link capacity and increase in packet loss. Specifically, the multiplexing size of 200 bytes gives the best compromise between capacity and packet loss for typical cellular applications.

This paper investigates the performance of packet networks carrying many low bit-rate real-time sources of traffic. We are primarily motivated by applications to the design of wide-area transmission links carrying coded voice traffic between the base sites and the core network of a cellular system. The large number of such links and distances they span lead to significant recurring costs for cellular operators forcing them to operate these links at very high utilization. The system model is composed of a highly utilized transmission link fed by a large number of voice sources. The voice sources are characterized by their correlation structure and relative time-offset with respect to each other. We present simulation and analytical results for this system emphasizing the impact of system utilization, correlation structure, etc. The analysis is based on two different approaches (1) Gaussian queue analysis and (2) heavy traffic analysis. The analysis reflects the statistical properties of the voice sources, in particular, their correlation structure. A key analytical result is that the delays in the system can be characterized by two scaled exponential distributions: one for small delays and the other for large delays.

We consider a model comprising of a single bottleneck node/switch fed by multiple Markov modulated fluid sources and with significant and possibly different propagation delays. The amount of traffic released by the sources depends on its state as well as delayed feedback from the network. Such a source model allows us to capture ABR as well as VBR traffic types. We obtain analytical results for computing various performance measures associated with this model. Based on these analytical results, we study the performance of the feedback policy for two different delay classes. In the parameter regime investigated, we observe that while the sources with a smaller delay do better, they do not do significantly better. Moreover, the performance of the farther sources improves as the second set of sources is moved closer. Thus, what may be perceived as unfairness due to the difference in feedback delays should really be interpreted as the closer sources helping the farther sources while helping themselves.

We consider a system comprising of a single bottleneck switch/node that is fed by N independent Markov-modulated fluid sources. There is a fixed propagation delay incurred by the traffic between these sources and the switch. We assume that the switch sends periodic feedback in the form of a single congestion indicator bit. This feedback also incurs a fixed propagation delay in reaching the sources. Upon reaching the sources (or the access controllers associated with the sources), this congestion indicator bit is used to choose between two rates for the excess traffic, high or low, possibly depending on the state of that source. The switch employs a threshold mechanism based on its buffer level to discard excess traffic. We show that the stationary distribution of this system satisfies a set of first-order linear differential equations along with a set of split boundary conditions. We obtain an explicit solution to these using spectral decomposition. To this end we investigate the related eigenvalue problem. Based on these results we investigate the role of delayed feedback vis-a-vis various time-constants and traffic parameters associated with the system. In particular, we identify conditions under which the feedback scheme offers significant improvement over the open-loop scheme.>